In contrast to lossy audio coding techniques (like mp3, AAC etc.), lossless compression algorithms can only exploit redundancies of the original audio signal to reduce the data rate. It is not possible to rely on irrelevancies, as identified by psycho-acoustical models in state-of-the-art lossy audio codecs. Accordingly, the common technical principle of all lossless audio coding schemes is to apply a filter or transform for de-correlation (e.g. a prediction filter or a frequency transform), and then to encode the transformed signal in a lossless manner. The encoded bit stream comprises the parameters of the transform or filter, and the lossless representation of the transformed signal. See, for example, J. Makhoul, “Linear prediction: A tutorial review”, Proceedings of the IEEE, Vol. 63, pp. 561-580, 1975, T. Painter, A. Spanias, “Perceptual coding of digital audio”, Proceedings of the IEEE, Vol. 88, No. 4, pp. 451-513, 2000, and M. Hans, R. W. Schafer, “Lossless compression of digital audio”, IEEE Signal Processing Magazine, July 2001, pp. 21-32.
The basic principle of lossy based lossless coding is depicted in FIG. 12 and FIG. 13. In the encoding part on the left side of FIG. 12, a PCM audio input signal SPCM passes through a lossy encoder 81 to a lossy decoder 82 and as a lossy bit stream to a lossy decoder 85 of the decoding part (right side). Lossy encoding and decoding is used to de-correlate the signal. The output signal of decoder 82 is removed from the input signal SPCM in a subtractor 83, and the resulting difference signal passes through a lossless encoder 84 as an extension bit stream to a lossless decoder 87. The output signals of decoders 85 and 87 are combined 86 so as to regain the original signal SPCM.
This basic principle is disclosed in EP-B-0756386 and U.S. Pat. No. 6,498,811, and is also discussed in P. Craven, M. Gerzon, “Lossless Coding for Audio Discs”, J. Audio Eng. Soc., Vol. 44, No. 9, September 1996, and in J. Koller, Th. Sporer, K. H. Brandenburg, “Robust Coding of High Quality Audio Signals”, AES 103rd Convention, Preprint 4621, August 1997. In the lossy encoder in FIG. 13, the PCM audio input signal SPCM passes through an analysis filter bank 91 and a quantisation 92 of sub-band samples to a coding and bit stream packing 93. The quantisation is controlled by a perceptual model calculator 94 that receives signal SPCM and corresponding information from the analysis filter bank 91.
At decoder side, the encoded lossy bit stream enters a means 95 for de-packing the bit stream, followed by means 96 for decoding the subband samples and by a synthesis filter bank 97 that outputs the decoded lossy PCM signal SDec.
Examples for lossy encoding and decoding are described in detail in the standard ISO/IEC 11172-3 (MPEG-1 Audio).
The two or more different signals or bit streams resulting from the encoding are to be combined so as to form a single output signal. Similar solutions exist for example for MPEG Surround, mp3PRO and AAC+. For the two latter examples the additional amount of data (SBR information) to be added to the base layer data stream (AAC or mp3) is small. Therefore this additional information can be packed into a standard-conform AAC or mp3 bit stream e.g. as ‘ancillary data’. Although the additional amount of data for the surround information is bigger than that for the SBR information, these data can still be packed into a standard-conform bit stream in the same way.
Another application using similar techniques is the ID3 tag added to mp3 standard audio streams. The data is added at the beginning or end of the existing mp3 file. A special mechanism is used so that an mp3 decoder does not try to decode this additional information.